;> What does this exactly mean and why am I receiving this ?
>> Beside re-enabling my subscription, what can I do to improve things ?
>>
>> Regards
>>
>>
>>
--
Rusty Newton
Digium, Inc
Removed from the list.
On Sun, Mar 19, 2017 at 11:54 AM, kenc <k...@vipmarketing.org> wrote:
> Dear friend!
>
>
>
> There is something really wonderful I wanted to show you, I hope you'll love
> that stuff) Please take a look read more
--
Rusty Newton
Digium, Inc. | C
you are safe to go ahead and file an issue report. Please
include the sip.conf/pjsip.conf plus a packet capture and Asterisk
debug log (be sure to get the DEBUG channel turned on in logger.conf)
with correlating SIP trace.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Da
n account on signup.asterisk.org.
Hope all that helps! Thanks.
[1]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
[2]: https://issues.asterisk.org/jira/secure/Dashboard.jspa
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 358
risk 13 which
is the most recent LTS.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: htt
acker, as it
> requires a client certificate for which I do not have.
> I'm running Asterisk 1.8.28.2; on Linux, CentOS 6.7 32-bit.
> Any help that can be provided would be appreciated.
Looks like you posted twice in the same day. I responded to your other
thread. Try not to duplicate pos
ten => s-CHANUNAVAIL,n,GotoIf($["${number}" = "9"]?106)
> exten => s-CHANUNAVAIL,106,SoftHangup(${EXTEN})
I suppose by bridgeConference you mean ConfBridge?
If you require assistance you'll need to describe more than what you
*want to do*. You'll need to describe the iss
p:sip.itco.nl
> client_uri=sip:tr...@sip.itco.nl
> contact_user=tryba
> outbound_auth=tryba_auth
> expiration=180
>
For those wandering web-searching souls:
https://issues.asterisk.org/jira/browse/ASTERI
documentation for that function is available at the CLI "core show
function REGEX" and is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_REGEX
It should be able to handle typical regular expression. I don't see
anything wrong with what you are doi
://lists.digium.com/mailman/listinfo/asterisk-users
This user has been removed from the list.
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Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256
is going on.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Chec
On Mon, Aug 24, 2015 at 9:40 PM, Pete Mundy p...@fiberphone.co.nz wrote:
Any chance the list admins could unsubscribe Mr Anzaldi until he gets his
broken auto-responder fixed?
He has been unsubscribed and alerted to the issue. Thanks!
--
Rusty Newton
Digium, Inc. | Community Support Manager
use both.. you will want to make sure your bind addresses and ports
don't conflict.
Why not use chan_pjsip for all SIP connectivity?
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - USdirect: +1 256 428 6200
Check us out at: http
://wiki.asterisk.org/wiki/display/AST/Logging
Once you have verbose output going to a log, make sure it is turned up to 5
and then post the call output to the list. With that we'll be able to see
what is happening.
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL
On Fri, Jul 17, 2015 at 9:09 AM, Luca Bertoncello lucab...@lucabert.de
wrote:
Rusty Newton rnew...@digium.com schrieb:
Perhaps the incoming calls are routed through different dialplan and in
that Dial you do not have the proper options? The dialplan you posted
appears to be for dialing
running into?
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - USdirect: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
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causecode in HANGUP
application because this can confuse a calling equipment.
I only know of the SIPAddHeader application which lets you add headers when
used before Dial, so I don't think you can do this currently.
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive
.
JR
Howdy, please don't cross-post onto the asterisk-users list with job
postings. They are allowed on asterisk-biz, but not on asterisk-users.
http://www.asterisk.org/community/discuss
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806
?
In Chapter 18 of Asterisk the Definitive Guide there is a section on LDAP
integration that might be helpful.
If you find any errors in wiki documentation please comment on the wiki
page or else file a bug report to let us know.
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis
messages then that would be helpful.
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - USdirect: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
probably want to read through the logging documentation on the wiki.
https://wiki.asterisk.org/wiki/display/AST/Logging
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Thanks,
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL
be a hassle. I had to turn it off on my home ASUS RT-N56U as
there was no configuration for it (it was even undocumented! yay!).
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - USdirect: +1 256 428 6200
Check us out at: http://digium.com http
configuration, or a bug.
If you can pastebin a full (sanitized) pjsip.conf as well as an Asterisk
log with verbose turned up[1], plus a SIP packet trace then we can take a
look at it.
[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Rusty Newton
Digium, Inc
On Thu, Apr 2, 2015 at 11:07 AM, Trey Hilyard kct...@gmail.com wrote:
I actually got the issue resolved by upgrading to 13.3.rc-1, since this
is just my development system. I assume that the problem was resolved
between the two releases.
Sweet, glad to hear!
--
Rusty Newton
Digium, Inc
the call without any
issue
how can ido in order to use DID in route inboud i use elastix
The best place to ask a question about Elastix configuration is the Elastix
forums, http://forum.elastix.org/.
The log output you show isn't enough to indicate the issue from what I can
see.
--
Rusty
have that, provide a pastebin link to the output and someone may
be able to help you out.
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - USdirect: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
to
Analog adapter that also communicated with Asterisk over IP.
Connecting a cell phone directly to Asterisk is a whole different
story. Do you mean an actual cellular device such as a GSM cell phone?
Or do you mean a phone that connects to a 802.11 Wi-Fi signal?
--
Rusty Newton
Digium, Inc
On Tue, Dec 23, 2014 at 6:34 PM, Rusty Newton rnew...@digium.com wrote:
On Tue, Dec 23, 2014 at 4:17 PM, Joseph syscon...@gmail.com wrote:
Are there any adapters that would allow me to connect asterisk to wifi or we
are not there yet?
I have Digium adapter S101i that was discontinued
.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth and Colocation Provided
/extension/priority after a blind transfer (use ^ characters in
place of | to separate context/extension/priority when setting this
variable from the dialplan)
Try using ^ characters as it mentions there.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL
a difference.
You probably want to file it as a bug.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
As the guidelines mention, be sure to include Asterisk logs with debug
showing the issue.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville
transfers on this site, and no other sites and you only started having
the blind transfers 7 months into usage.
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Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
wanted to make sure a link to this
relevant issue was here for the archives:
https://issues.asterisk.org/jira/browse/ASTERISK-24596
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com
of unclear at the moment unfortunately -
https://issues.asterisk.org/jira/browse/ASTERISK-24596
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
documentation on the wiki.
I haven't tried it myself in a long while, however Google was supposed
to end XMPP support for GV back in May.
I've heard mixed reports from community members.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct
://community.freepbx.org/
Thanks,
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth
/Collecting+Debug+Information
You might also mention the exaction version of Asterisk you are using
and which channel driver (though it sounds like chan_sip based on the
options described).
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
/telephony-cards/digital/single-span
Your question isn't very specific, so if you are having problems with
dialplan you will need to elaborate on Asterisk version, hardware
drivers and Asterisk channel drivers in use, etc.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW
think there is a *simple* way to do it.
If you are using external scripts with Asterisk APIs such as AMI or
ARI, then you can probably accomplish what you want using those
interfaces.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
+Commands
[2]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Command+Reference
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
this in a DB table for it to be
expressed correctly? I'm using MySQL.
There is an existing report filed here:
https://issues.asterisk.org/jira/browse/ASTERISK-19254
You can try Walter's suggestion on the issue and report back whether
it works or not.
--
Rusty Newton
Digium, Inc. | Community Support
. That is according to the
documentation.. which every once in a while is wrong. Other than that,
it should record as long as the channel is bridged.
Can you pastebin a log showing that particular call?[1]
[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Rusty Newton
on a duplicate topic so
quickly. It can create noise in the list and cause confusion when some
people reply to your second thread vs your first. I've replied to your
first thread.
Re: maxduration, I believe that is an option for Record and not
MixMonitor, but I could be wrong.
Thanks,
--
Rusty Newton
.
* The steps you took to produce the issue.
* What you expected to happen and what actually happened.
* What version of Zoiper you are using.
* The destination of the SIP text message you are trying to send it
to. (i.e. is this phone to phone, or phone to Asterisk, etc).
Thanks,
--
Rusty Newton
for one reason or another.
For anyone to help you, you would probably need to post your sip.conf
configuration (full) and screenshots of your soft/hard client
configuration.
Obviously you would want to use a fake password and sanitize IP
addresses if necessary.
Thanks,
--
Rusty Newton
Digium
this?
There is a tutorial for secure calling with TLS and SRTP here:
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com
Channels, let us know.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth
)
;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
; unauthenticated sessions that will be allowed
; to connect at any given time. (default: 100)
--
Rusty Newton
Digium, Inc. | Community Support Manager
445
.
If commands like core show channel channel and sip show channel
channel work then you'll want to attach that data as well.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http
checked and neither does certified Asterisk 1.8.15?
Thanks for taking note. I've filed an issue here
https://issues.asterisk.org/jira/browse/ASTERISK-24104
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us
gcc (GCC) 4.8.3 20140624 (Red Hat 4.8.3-1)
many thanks
Philip
I've rarely seen a machine that it isn't available on. The exception
for me was a virtualbox machine in one particular case.
You may get more help if you describe more about the CPU/architecture
that the machine uses.
--
Rusty
duplicate threads in such a short time span. You
posted this already yesterday. You'll have to be patient and wait for
someone to respond.
Thanks,
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - USdirect: +1 256 428 6200
Check us out
may be able to help you.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
to use quotes with the
full path. I cannot remember this second.
Try
#include /etc/asterisk/pjpeers.conf
Also it is funny to use peers in the file name since there is no
peers concept in res_pjsip. :)
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville
documentation at
https://wiki.asterisk.org/wiki/display/AST/Directory+and+File+Structure
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
the
username in.
If this wasn't FreePBX I'd tell you to just try setting the callerid
and fromuser options for the corresponding SIP peer. I don't want to
pretend to know FreePBX, so I still recommend you go ask on their
forum to get better assistance.
Good luck!
--
Rusty Newton
Digium, Inc
and Asterisk
Manager Interface
http://shop.oreilly.com/product/0636920025894.do
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
Hi! Please use the asterisk-biz mailing list for commercial/paid
support/job posting type discussions.
http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Simply use a hard phone instead of a soft-phone. Then go from there on
to two phones.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
On Fri, May 30, 2014 at 7:42 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hi Rusty,
We found the problem - a configuration error. Thank you for the response.
I'm glad you found the issue! No problem.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW
the required schema?
https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
.
If you can reproduce the issue and provide debug to demonstrate, then
you might file a bug report.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
options.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth and Colocation
you are
doing.
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Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth
inside
sip.conf, Asterisk correctly detects a change in the included conf
file upon a sip reload.
You might try reproducing the issue on a fresh install, on your
non-production system to see if you can narrow down where the
difference is.
--
Rusty Newton
Digium, Inc. | Community Support Manager
On Wed, May 7, 2014 at 10:14 AM, Administrator TOOTAI ad...@tootai.net wrote:
Le 07/05/2014 16:50, Rusty Newton a écrit :
I contructed a basic sip.conf, and added this line to the end:
#include /etc/asterisk/sip_includes/*.conf
Here is the point. Modify it the way explained in previous
://issues.asterisk.org/jira/browse/ASTERISK-23683 and edited the
Summary and Description fields, as well as linked it an issue where
the fix for that issue *may* have introduced the problem you found.
Thanks!
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville
with two different included files in each
sip.conf and queues.conf, one in /tmp and one in /etc/asterisk. Same
working behavior.
I used SVN-branch-11-r413305, so you might want to test there.
However I'm still confused as to how you are seeing the behavior you
are seeing.
--
Rusty Newton
Digium
.
On a related note, the extra- set appears to have been converted from
the old GSM format. Are there any plans to have them re-recorded in
good quality?
Not at this time, you are welcome to contact Allison and ask her if
she would be willing to do that.
--
Rusty Newton
Digium, Inc. | Community
; the book I linked above contains some
great information on AGI and is written by some of our notable
community members. The latest edition is available here:
http://shop.oreilly.com/product/0636920025894.do
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW
interface cards that two B-channel transfers did something like this.
Digium has documentation on that here:
http://kb.digium.com/articles/Configuration/Two-B-Channel-Transfers
If that doesn't help, and you have a Digium card; you might call
Digium tech support to ask about it.
--
Rusty Newton
Digium
into Asterisk
variables.. there may be other ways to do what you want.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
-11.9.0-rc2.tar.gz
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth
to the console, or
else possibly with output from rtp set debug on.
As for the show channels type commands, it may say something about
encryption rather than SRTP directly. I'll take a look later if I
get a chance.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW
On Mon, Mar 31, 2014 at 1:26 PM, Rusty Newton rnew...@digium.com wrote:
On Fri, Mar 28, 2014 at 7:39 PM, Patrick Laimbock patr...@laimbock.com
wrote:
Hi,
I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI
command to see if SRTP is active on a channel/call. I went through
then show that configuration as well. Based on that someone
could make a suggestion.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
Which looks like the same issue that you are having. If you click on
the source tab you can see the commits it was fixed in. Looks like it
was after 11.8.1, so you'll have to wait until the next release, or
grab 11 from SVN.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis
/ASTERISK-21384
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth
is not a reason to repeatedly
post (four posts now in the past few days?)
I've already responded to your original post and asking you to post on
the issue tracker and follow the issue guidelines to provide the
information needed to investigate the crash.
--
Rusty Newton
Digium, Inc. | Community Support
|
+-+-+-+-+---++-+--+-+-+--+
What is clid 100 100? Why it came from? No this source into log.
That is the Caller ID information for that channel.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive
. Be sure to provide instructions
and configuration that would allow us to reproduce the issue.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
/display/AST/Asterisk+11+ManagerEvent_ConfbridgeEnd
That is all I got from poking around the docs. :)
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
...
exten = 1,n,Set(CONFBRIDGE(user,record_file)=${MyCustomFileName}.wav)
Then of course you now know the file name so you could do whatever you
wanted with it afterwards.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CONFBRIDGE
--
Rusty Newton
Digium, Inc. | Community Support
to configure Asterisk to hang up during
the ringing phase when a peer/endpoint becomes unreachable. I don't
see an option or parameter for that behavior.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http
detect when two users are
behind the same NAT and redirect their traffic inside that NAT; this way the
load of RTP traffic on Asterisk server will be reduced.
I don't know that this is possible with any simple Asterisk
configuration. Good luck!
--
Rusty Newton
Digium, Inc. | Community Support
is already on the wiki.
Here https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance+Basics
and here https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance
Those two pages and their sub-pages have some overlap and may need to
be consolidated.
--
Rusty Newton
Digium, Inc. | Community
with this option before, but it sounds like the
intent is what you may need.
A link to the sample file (that is also included with your source
files)
http://svnview.digium.com/svn/asterisk/branches/11/configs/sip.conf.sample?view=markup
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan
me or someone in #asterisk-dev know and we'll make
sure things get updated. One thing I do have on my to-do list is a NAT
guide.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com
on in their network, receiving it
back at 5060 is no guarantee it'll get back to your Asterisk VM.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
the previously caller directly.
Hope any of that helps. *Goes off to document SIPFROMDOMAIN and
SIPFROMUSER on the wiki*
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http
Actually SIPFROMDOMAIN was documented here:
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
, but SIPFROMUSER was not. They are now both there! :)
On Wed, Feb 19, 2014 at 9:08 AM, Rusty Newton rnew...@digium.com wrote:
On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
/display/AST/Asterisk+Versions
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
/Configuring+res_pjsip
[3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
Hope that helps, thanks!
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
On Thu, Jan 23, 2014 at 7:01 PM, Rusty Newton rnew...@digium.com wrote:
snip
the 1.8,11, or 12 branches. That being said, 12 is rather new and has
many significant changes that should be considered.[3]
I meant to reference link [1] of course. :)
--
Rusty Newton
Digium, Inc. | Community
/viewtopic.php?p=195944
It won't use MD5. It only uses Basic.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
+ManagerAction_Redirect
https://wiki.asterisk.org/wiki/display/AST/AMI+Examples
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
.
Does it ring them in the same order for every call(even if it is not
the expected order)?
Does the order change up for each call even when no new members have been added?
Can you provide a pastebin of a verbose log (see logger.conf)
demonstrating the problem?
--
Rusty Newton
Digium, Inc
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
://lists.digium.com/mailman/listinfo/asterisk-users
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
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